hello all - got a question - I've only recently been stopping and taking a
look at my studio computer's performance and in the almost year since I
change from Red Hat 9 to gentoo, it's been more solid on some things, but I
notice a huge latency difference - ie: I have to run Jack at -p 8192 to get
anything done in Ardour
Anybody have any tips on what to look at to tweak it? Seems like it should
do better than that... I didn't see it as a problem until in the last few
days I started playing with playing softsynths live directly into Ardour -
you've gotta be running at -p 1024 or there's a latency that screws up your
playing - at 8192 it's a downright 8th note delay...
Here's some vitals that I can think of:
OS: gentoo 2.6.6-rc1 kernel (alsa built in)
jack: 0.99.0
ardour: beta28
jack command line:
jackd -R -d alsa -d hw:0 -r 48000 -p 8192 <------- (or whatever)
harddrive:
multicount on
io support: 32 bit
unmaskirq on
use dma on
keepsettings off
readonly off
readahead on
chip: 2ghz amd (I THINK - not at computer now)
ram: 512MB
thanks for any ideas! :)
---------------------
Aaron Trumm
www.nquit.com
-----------------------
>From: Neil Durant <lists(a)sphere3.co.uk>
>
>I'd be happy to sample the full lengths of all 35 notes for all six sounds,
>storing them as wavs. I don't have a lot of spare time these days, so I'll
>let someone else have the pleasure of cropping/editing!
Please do so. Make the files available from your site or
upload to
ftp://ftp.funet.fi/incoming/audio/
Note: If we end up to conclusion that the samples cannot
be put freely available, then I could make the samples
privately available for the following kind of project.
Research experts should analyse the sounds and come up
with synthesis method which generates as similar sounds
as possible. I'm aware of such research teams and I could
ask them to analyse the samples.
I also have a plenty of research papers on such analysis/synthesis
methods. I could place the papers privately available for anyone
who wish to write the analysis/synthesis software.
Regards,
Juhana
--
http://music.columbia.edu/mailman/listinfo/linux-graphics-dev
for developers of open source graphics software
Hi Folks,
So, I have spouted off about a distro before (I know some of you checked
it out) and it was not as good as I hoped it would be! :(
Trust me...THIS is not the case this time!!!
Announcing PClinuxOS (PCLOS for short)
http://www.pclinuxonline.com/pclos/
This is based on Mandrake but minus the bloat! One ISO built as a
LiveCD! You can boot it and not install...no commitment unless you want
to. The install is simple as can be and fast. KDE 3.4, Fluxbox, Gnome,
etc. The distro was started by Texstar who used to build packages for
Mandrake a few years ago.
So, whats that got to do with Music?? Just this; Thac has decided to
start packaging for PCLOS just as he has with Mandrake. For those not
familiar with his work...
http://rpm.nyvalls.se/sound10.1.html
I have relied on Thac's RPMS for audio since I started Linux audio work.
He keeps his packages updated and has about everything worth having in
his repositories. They have been invaluable to me.
As recently as yesterday, Thac has started his own 3rd party Apt
repository for PCLOS and as I write I am downloading the first of his
packages and new mm kernel, etc. We are chatting on the Pclinuxos IRC
chat room and ironing out a few minor bugs. It will be a few days before
he has all of his packages done.
Texstar suggested that we might make a liveCD with the key audio apps
and a R/T kernel for easy usage and evaluation. Obviously some time is
needed to work the bugs out but this is very likely in the not too
distant future! I should think those of you working with MDK right now
would see this as an easy step....others...well, the proof will be in
the pudding.
More soon....
R~
I don't think I understand this jackplug concept.
I'm trying to get an alsa client, such as aplay to play through jackd.
I've setup jackplug in /etc/asound.conf
pcm.jackplug
{
type plug
slave
{
pcm "jack"
}
}
pcm.jack
{
type jack
playback_ports
{
0 alsa_pcm:playback_1
1 alsa_pcm:playback_2
}
capture_ports
{
0 alsa_pcm:capture_1
1 alsa_pcm:capture_2
}
}
I then tell aplay to use:
#aplay -d jackplug foobar.wav
But aplay plays the file even if jackd is not started; why?. Isn't
jackplug supposed to show up in my jack connections when aplay is
playing?
I seem to be missing something vital.
Is there some way to make a "fake audio device", like hw:3,0 that alsa
applications can connect to so that I can get the audio out and then
send it through jack, process it and then send it to the real audio
device?
--
Esben Stien is b0ef(a)esben-stien.name
http://www.esben-stien.name
irc://irc.esben-stien.name/%23contact
[sip|iax]:b0ef@esben-stien.name
Hi,
finally the Wiki went online:
http://wiki.freeaudiosoftware.orghttp://freeaudiosoftware.org currently is a simple redirect to the wiki.
About this Wiki: The idea was that we have a Wiki to collect hints, tips
and solutions we see in this list.
Why a new domain?
I often found useful hints in this list, but none of the existing wikis
seemed to be on topic, so I stored info on my private machine only.
It's a pity.
So if you get valuable info from this list or want to contribute to the
wiki in any way I'd be glad if the Wiki grew to a central place where
users of free audio software will get a good starting point to get info
about tutorials, software usage, an overview over important web pages
and so on.
I hope you like it and help making it worth the efforts.
Best regards
ce
I have a Sony CD ROM in my Linux box. It has worked fine from the
beginning but around the time I recently moved from Fedora Core 1 to
Core 3 (Planet CCRMA), it stopped working for burning (using xcdroast)
and playing through my Delta 1010LT sound card. The timing could be a
coincidence but I suspect the setup got messed up somehow in the
transition. If I put a audio CD in it, the sound comes out fine if I
plug my headphones into the built-in (to the player) headphone jack but
no sound comes through my sound card.
My sound card is working fine for other apps like Audacity, Ardour,
aplay, etc.
Can someone help me figure out what might have gone wrong?
Thanks in advance for any tips.
Mike
Mike Jewell
Me again.
More music: http://blog.dis-dot-dat.net/2005/08/stress-and-music.html
Also includes sample breakdown.
Why not take the same samples and do something different? Would be
interesting...
James
--
"I'd crawl over an acre of 'Visual This++' and 'Integrated Development
That' to get to gcc, Emacs, and gdb. Thank you."
(By Vance Petree, Virginia Power)
Part 2 of (maybe) 3
For those with lack of familiarity with Fourier analysis and
synthesis, here is a concrete example to demonstrate potentially
serious problems with sinc resamplers in doing bulk conversions at
constant rates. These problems are real and could easily result in
audible artifacts --- something that I assume is of importance
to Linux audio users --- and especially with further processing.
------------------------
File 123163main_cas-skr1-112203.wav is the NASA file recently
mentioned on LAU --- a public-domain, taxpayer-supported WAV file
sampled at 5000 samples per second. This file was chosen arbitrarily
--- just happened to resample it before reverbing and posting for
interested LAU'ers a while ago, so decided to use it for a
comparison for Steve Harris.
Two resamplers:
1) sinc resampler:
$ sndfile-resample -to 44100 -c 0 123163main_cas-skr1-112203.wav \
saturn_sndfile-resample.wav
2) FFT with large windows:
Sampster in Mixster (stuff I wrote myself)
Comparison was every 50th sample in the original file with every 441st
sample in the other two (should match exactly every 0.01 seconds) for
the first 9.5 seconds of the files. 9.5 seconds was chosen rather
arbitrarily --- nothing special about it. Ideally these particular
samples should match exactly. Any error indicates corruption of the
original data at the exact locations where the original samples were
taken. The last two columns show you the difference between what is
expected at these matching points and what was actually obtained after
resampling. Note that the values in the last column are significantly
greater than those in the next-to-last column.
Match# Original FFT sndfile # FFT sndfile (diffs)
1: 0.00000 -2.00000 19.0000 1: -2 19
2: 386.000 384.000 437.000 2: -2 51
3: -181.000 -183.000 -178.000 3: -2 3
4: -500.000 -502.000 -538.000 4: -2 -38
5: -1065.00 -1067.00 -1068.00 5: -2 -3
6: -54.0000 -56.0000 -28.0000 6: -2 26
7: -120.000 -122.000 -55.0000 7: -2 65
8: -348.000 -350.000 -344.000 8: -2 4
9: 827.000 825.000 805.000 9: -2 -22
<snip>
344: -67.0000 -71.0000 100.000 344: -4 167
345: -378.000 -382.000 -275.000 345: -4 103
346: -37.0000 -41.0000 -101.000 346: -4 -64
347: -209.000 -213.000 -19.0000 347: -4 190
348: 269.000 265.000 86.0000 348: -4 -183
349: 62.0000 58.0000 27.0000 349: -4 -35
350: 427.000 423.000 446.000 350: -4 19
351: 154.000 150.000 -47.0000 351: -4 -201
352: 619.000 615.000 52.0000 352: -4 -567
353: -202.000 -206.000 111.000 353: -4 313
354: -366.000 -370.000 205.000 354: -4 571 <<<
OUCH! Hope this doesn't get expanded. Over 100x larger error.
355: -146.000 -150.000 8.00000 355: -4 154
356: 549.000 545.000 558.000 356: -4 9
357: 279.000 275.000 -34.0000 357: -4 -313
358: -110.000 -114.000 -12.0000 358: -4 98
359: -184.000 -188.000 199.000 359: -4 383
360: -215.000 -219.000 -417.000 360: -4 -202
361: 244.000 240.000 74.0000 361: -4 -170
362: -474.000 -478.000 -152.000 362: -4 322
363: 188.000 184.000 562.000 363: -4 374
<snip>
938: -1448.00 -1449.00 -1468.00 938: -1 -20
939: -1203.00 -1204.00 -1161.00 939: -1 42
940: 3210.00 3209.00 3111.00 940: -1 -99 <<< about
100x larger error at 10% full scale
941: 5767.00 5766.00 5838.00 941: -1 71
942: -656.000 -657.000 -628.000 942: -1 28
943: -5165.00 -5166.00 -5163.00 943: -1 2
944: 1547.00 1546.00 1584.00 944: -1 37
945: 4410.00 4409.00 4445.00 945: -1 35
946: 1912.00 1911.00 1881.00 946: -1 -31
947: 5947.00 5946.00 5829.00 947: -1 -118 <<< Over
100x larger error at 18% full scale.
948: 5923.00 5922.00 5902.00 948: -1 -21
949: 3462.00 3461.00 3494.00 949: -1 32
What this shows is that at every 0.01 seconds, where the original file
and the resampled file should have the same exact value (if the
original data were preserved), large errors occur for sndfile-
resample.
------------------------
resample-1.7 was even worse with a phase shift on top of this type of
inaccuracy, coupled with rather serious spectral leakage beyond 2.5
kHz which was the original band limit (or might as well be assumed to
have been). Upon examining the waveforms, I could see that resample-1.7
was doing an excellent job of tracing out the original waveform by
drawing pretty much straight lines between points. Although visually
reassuring, this actually adds spectral components that were not
in the original. So it depends on what you want. This resampling
probably won't sound like the original, but does look good in an
editor.
------------------------
Also of interest is that the very latest version of sndfile-resample
gives slightly different results than an earlier version for the
locations which should match (the versions are for libsamplerate):
Match# v 0.0.15 v 0.1.2
1: 19.0000 18.0000
2: 437.0000 436.0000
3: -178.0000 -179.0000
4: -538.0000 -539.0000
5: -1068.0000 -1068.0000
6: -28.0000 -28.0000
7: -55.0000 -55.0000
8: -344.0000 -344.0000
9: 805.0000 805.0000
10: -81.0000 -82.0000
11: 482.0000 482.0000
12: 78.0000 77.0000
13: 227.0000 227.0000
14: 501.0000 500.0000
15: 13.0000 12.0000
<snip>
So the *amount* of corruption of the original data at locations which
should match varies with version! Fortunately (or perhaps unfortunately
depending upon your point of view) this latest version never varies
more than 2 from the earlier version, so the "latest and greatest" is
just as bad. The errors in the table above would be altered by 2 or
less, which is insignificant.
I've discovered the international symbol for the LAU Illuminati on, of all
places, my very own web page. And I'm just a lurker! How dastardly! At any
rate, here it is.
http://towndowner.com/lau-illuminati.jpg
well, I thought it was funny. I'm a terrible photoshopper, and I've been
reading way too much fark/somethingawful.
this mailing list and its participants are fantastic - i read and learn from you
guys every day - thank you. keep illuminatizing!
--
dan easley (dan(a)burntpossum.com)
prabob